[asterisk-dev] [Code Review] Pimp My SIP Media Improvements

[jcolp]

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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/2318/
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(Updated Feb. 11, 2013, 6:39 a.m.)


Review request for Asterisk Developers.


Changes
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Fixed a bug with RTP instance destruction uncovered during some testing.


Summary
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These changes clean up media handling, move some more stuff into res_sip_sdp_audio, fixes a few bugs, and adds some additional features.

The act of negotiating an SDP media stream and actually applying the media stream are now separate operations.
Hold/unhold works.
RTP over IPv6 works.
Use of the 'ptime' attribute works.
Local Packet2Packet bridging works.
Symmetric RTP can now be enabled per-endpoint.
Reduced memory pool usage.
Fixed bug where the RTP instance was never destroyed.


Diffs (updated)
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  /team/group/pimp_my_sip/channels/chan_gulp.c 381157 
  /team/group/pimp_my_sip/configs/res_sip.conf.sample 381157 
  /team/group/pimp_my_sip/include/asterisk/res_sip.h 381157 
  /team/group/pimp_my_sip/include/asterisk/res_sip_session.h 381157 
  /team/group/pimp_my_sip/res/res_sip/sip_configuration.c 381157 
  /team/group/pimp_my_sip/res/res_sip_sdp_audio.c 381157 
  /team/group/pimp_my_sip/res/res_sip_session.c 381157 

Diff: https://reviewboard.asterisk.org/r/2318/diff


Testing
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1. Sent and received calls from a few different devices
2. Held/unheld a call
3. Attempted to set up incompatible calls (only configured for gsm, but offering ulaw only)


Thanks,

jcolp

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